[Freeswitch-users] : need help on call to gateway
false
cucku.cucku at yahoo.com.vn
Thu Apr 1 07:17:48 PDT 2010
Hi Francios
Thank you for you recommend
from your recommendation, i add the host name : sip.yeah.com into the
/etc/hosts
but i am still getting error on DNS
Failed with status DNS Error [503]
it seems that FS needs resolve the sip.yeah.com from the DNS server.
the FS support register with domain - domain is not real and the parameter = register-proxy. Is there the same paramater for make call out??
i do ping sip.yeah.com successfull but when i do nslookup and the DNS server response cannot find the domain name
[root at localhost external]# nslookup sip.yeah.com
Server: 8.8.8.8
Address: 8.8.8.8#53
** server can't find sip.yeah.com: NXDOMAIN
[root at localhost external]# ping sip.yeah.com
PING sip.yeah.com (118.69.239.250) 56(84) bytes of data.
64 bytes from sip.yeah.com (118.69.239.250): icmp_seq=2 ttl=250 time=6.84 ms
Thank you
Ha`
________________________________
Từ: François Legal <devel at thom.fr.eu.org>
Đến: freeswitch-users at lists.freeswitch.org
Gửi ngày: 20:51:17, Thứ Tư, 31 tháng 3 2010
Chủ đề: Re: [Freeswitch-users] Về: Về: Về: need help on call to gateway
Then either provide DNS to that system so that it can resolve the address, either create a static binding in /etc/hosts file on that host for sip.yeah.com
On Wed, 31 Mar 2010 20:36:37 +0800 (SGT), false <cucku.cucku at yahoo.com.vn> wrote:
Hi David
>
>when change the
>the Register message send out with wrong format : from sip:071234 at x.x.x.x
>the right format : sip:071234 at sip.yeah.com
>
>is there any way to fix it
>
>Thank you
>
>
>
>
>
________________________________
Từ: David Ponzone
>Đến: freeswitch-users at lists.freeswitch.org
>Gửi ngày: 15:50:00, Thứ Tư, 31 tháng 3 2010
>Chủ đề: Re: [Freeswitch-users] Về: Về: need help on call to gateway
>
>I guess you need to put the IP of the proxy also in:
>
>
>David Ponzone Direction Technique
>email: david.ponzone at ipeva.fr
>tel: 01 74 03 18 97
>gsm: 06 66 98 76 34
>
>
>Service Client IPeva
>tel: 0811 46 26 26
>www.ipeva.fr - www.ipeva-studio.com
>
>
>Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
>
>
>
>
>
>Le 31/03/2010 à 10:22, false a écrit :
>
>Hi David
>>
>>the domain sip.yeah.com cannot resolve in the FS host
>>the thing here is FW support for Registration to remote SIP server
>>so the domain no need to resolve
>>how to do the same thing when call out, no need to resolve the domain name: sip.yeah.com,
>>
>>Thank you
>>
>>
>>
>>
________________________________
Từ: David Ponzone <david.ponzone at gmail.com>
>>Đến: freeswitch-users at lists.freeswitch.org
>>Gửi ngày: 15:12:50, Thứ Tư, 31 tháng 3 2010
>>Chủ đề: Re: [Freeswitch-users] Về: need help on call to gateway
>>
>>This looks like a DNS error.
>>
>>Are you sure you can resolve sip.yeah.com from the FS host ?
>>
>>
>>David Ponzone Direction Technique
>>email: david.ponzone at ipeva.fr
>>tel: 01 74 03 18 97
>>gsm: 06 66 98 76 34
>>
>>
>>Service Client IPeva
>>tel: 0811 46 26 26
>>www.ipeva.fr - www.ipeva-studio.com
>>
>>
>>Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
>>
>>
>>
>>
>>
>>Le 31/03/2010 à 09:51, false a écrit :
>>
>>Hi David and Brian
>>>
>>>sorry to bother you again
>>>the domain name : sip.yeah.com is not real, How to fix the issue
>>>
>>>FreeSwitch register to sip server with user 071234 successull
>>>FreeSwitch receive incomming from sip server and then transfer to 1000 extension successfull
>>>the extension make call out using the sip server is fail
>>>and the error :
>>>sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
>>>nta: for "sip.yeah.com" query "sip.yeah.com" A (cached)
>>>nua(0x9473108): call state changed: init -> calling, sent offer
>>>soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c], [(nil)]) called
>>>nua(0x9473108): event i_state INVITE sent
>>>nua(0x9473108): event r_invite 503 DNS Error
>>>nua(0x9473108): call state changed: calling -> init
>>>nua(0x9473108): event i_state 503 DNS Error
>>>nua(0x9473108): event i_terminated 503 DNS Error
>>>
>>>
>>>i create 3 file:
>>> 1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/external for register to sip server
>>> 1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/default for outbound call
>>> 1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/dialplan/public for incomming call
>>>
>>>the content of 071234.xml :
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>the content of 00_inbound_did_fpt.xml:
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>the content of 01_fpt.net.xml:
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
________________________________
Từ: David Ponzone <david.ponzone at gmail.com>
>>>Đến: freeswitch-users at lists.freeswitch.org
>>>Gửi ngày: 14:26:13, Thứ Ba, 30 tháng 3 2010
>>>Chủ đề: Re: [Freeswitch-users] need help on call to gateway
>>>
>>>False,
>>>
>>>I think Brian is sleeping so I give you a quick answer.
>>>For OUTBOUND:
>>>Following my previous mail, you have to check if your user 1000 has the default context configured.
>>>If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/.
>>>(You may also remove from this path the remaining .xml files from the default install)
>>>In 01_whatever.xml, put:
>>>
>>>
>>>
>>>
>>>
>>>
>>>So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.
>>>For INBOUND:
>>>In conf/dialplan/public/, add a file named whatever.xml with the following content:
>>>
>>>
>>>
>>>
>>>
>>>
>>>Here, YOUR_DID is the DID number your provider allocated to you.
>>>That is an exact match, so if you're not sure about the exact format, remove the leading ^, and match that on the longest part of the number you know about.
>>>Of course, after doing all this, reloadxml.
>>>It will perhaps not fit in your config exactly, but you should get the idea to get started.
>>>
>>>
>>>David Ponzone Direction Technique
>>>email: david.ponzone at ipeva.fr
>>>tel: 01 74 03 18 97
>>>gsm: 06 66 98 76 34
>>>
>>>
>>>Service Client IPeva
>>>tel: 0811 46 26 26
>>>www.ipeva.fr - www.ipeva-studio.com
>>>
>>>
>>>Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
>>>
>>>
>>>
>>>
>>>
>>>Le 30/03/2010 à 09:01, false a écrit :
>>>
>>>Hi Brian
>>>>
>>>>could you guide me more to config on freeswitch
>>>>
>>>>i want xilte with user 1000 can make call out to sip server
>>>>and the prefix when make call out to sip server : 0
>>>>and receive incomming call from SIP server
>>>>
>>>>what file name i should create
>>>>
>>>>Thank you
>>>>
>>>>
>>>>--- Ngày Thứ 3, 30/03/10, Brian West <brian at freeswitch.org> đã viết:
>>>>
>>>>
>>>>>Từ: Brian West <brian at freeswitch.org>
>>>>>Chủ đề: Re: [Freeswitch-users] need help on call to gateway
>>>>>Đến: freeswitch-users at lists.freeswitch.org
>>>>>Ngày: Thứ Ba, 30 tháng 3, 2010, 4:37
>>>>>
>>>>>
>>>>>its looking in context default... not public.
>>>>>
>>>>>/b
>>>>>
>>>>>
>>>>>On Mar 29, 2010, at 11:33 PM, false wrote:
>>>>>
>>>>>Hi all
>>>>>>
>>>>>>i use freeswitch to register to sip server
>>>>>>the domain is not real: sip.yeah.com
>>>>>>the outbound proxy : 118.69.145.5
>>>>>>
>>>>>>i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
>>>>>>
>>>>>> sip.yeah.com">
>>>>>>
>>>>>>
>>>>>> sip.yeah.com"/>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>so i use xlite register to freeswitch with user 1000
>>>>>>i start freeswitch , freeswitch start ok,
>>>>>>freeswitch register to sip server successfull with username 071234
>>>>>>xlite register to freeswitch successfull
>>>>>>
>>>>>>so i edit the public.xml
>>>>>>for incomming call from sip server, i will forward to extension 1000
>>>>>> sip.yeah.com">
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>but when i use xlite make call out from freeswitch to sip server and get no routing
>>>>>>
>>>>>
>>>>________________________________
>>>>Tốt hơn, thoáng gọn hơn, nhanh hơn - Trải nghiệm Yahoo! Mail mới hôm nay!_______________________________________________
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>>>>
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